Spectrum coding/decoding apparatus and method for reducing distortion of two band spectrums

ABSTRACT

A coding apparatus capable of coding a spectrum at a low bit rate and with high quality without producing any disturbance in a harmonic structure of the spectrum. In this apparatus, internal state setting section ( 106 ) sets an internal state of filtering section ( 107 ) using first spectrum S 1 ( k ). Pitch coefficient setting section ( 109 ) outputs pitch coefficient T by gradually changing it. Filtering section ( 107 ) calculates estimated value S′ 2 ( k ) of second spectrum S 2 ( k ) based on pitch coefficient T. Search section ( 108 ) calculates the degree of similarity between S 2 ( k ) and S′ 2 ( k ). At this time, pitch coefficient T′ corresponding to the maximum calculated degree of similarity is given to filter coefficient calculation section ( 110 ). Filter coefficient calculation section ( 110 ) determines filter coefficient β i  using this pitch coefficient T′.

TECHNICAL FIELD

The present invention relates to a coding apparatus mounted on a radiocommunication apparatus or the like for coding a voice signal, audiosignal or the like and a decoding apparatus for decoding this codedsignal.

BACKGROUND ART

A coding technology for compressing a voice signal, audio signal or thelike to a low bit rate signal is particularly important from thestandpoint of effectively using a transmission path capacity (channelcapacity) of radio waves or the like and a recording medium in a mobilecommunication system.

Examples of a voice coding scheme for coding a voice signal includeschemes like G726, G729 standardized by the ITU-T (InternationalTelecommunication Union Telecommunication Standardization Sector). Theseschemes use narrow band signals (300 Hz to 3.4 kHz) as coding targetsand can perform high quality coding at bit rates of 8 kbits/s to 32kbits/s. However, since such a narrow band signal is so narrow that itsfrequency band is a maximum of 3.4 kHz, the quality thereof is such thatit gives the user an impression that a sound is muffled, which resultsin a problem that it lacks a sense of realism.

Furthermore, there is also a voice coding scheme that uses widebandsignals (50 Hz to 7 kHz) as coding targets. Typical examples of this areG722, G722.1 of ITU-T and AMR-WB of 3GPP (The 3rd Generation PartnershipProject). These schemes can perform coding of wideband voice signals ata bit rate of 6.6 kbits/s to 64 kbits/s. However, when the signal to becoded is voice, although a wideband signal has relatively high quality,it is not sufficient when an audio signal is the target or a voicesignal of higher quality with a sense of realism is required.

On the other hand, when a maximum frequency of a signal is generally onthe order of 10 to 15 kHz, it is possible to obtain a sense of realismequivalent to FM radio, and when the maximum frequency is on the orderof up to 20 kHz, it is possible to obtain quality comparable to that ofCD (compact disk). For such a signal, audio coding represented by thelayer III scheme or AAC scheme standardized by MPEG (Moving PictureExpert Group) is appropriate. However, these audio coding schemes have awide frequency band of a signal to be coded, which results in a problemthat the bit rate of a coded signal increases.

Examples of conventional coding technologies include a technology ofcoding a signal with a wide frequency band at a low bit rate (e.g., seePatent Document 1). According to this, an input signal is divided into asignal of a low-frequency domain and a signal of a high-frequencydomain, the spectrum of the signal of the high-frequency domain isreplaced by the spectrum of the signal of the low-frequency domain andcoded, and the overall bit rate is thereby reduced.

FIG. 1A to FIG. 1D show an overview of the above described processing ofreplacing the spectrum of high-frequency domain by the spectrum of thelow-frequency domain. This processing is originally intended to beperformed in combination with coding processing, but for simplicity ofexplanation, a case where the above described processing is performed onan original signal will be explained as an example.

FIG. 1A shows a spectrum of an original signal whose frequency band isrestricted to 0≦k<FH, FIG. 1B shows a spectrum of the signal restrictedto 0≦k<FL (where, FL<FH), FIG. 1C shows a spectrum obtained by replacinga high-frequency domain (high-frequency band) by a low-frequency domain(low-frequency band) using the above described technology and FIG. 1Dshows a spectrum obtained by shaping the replacing spectrum according tospectrum envelope information about the replaced spectrum. In thesefigures, the horizontal axis shows a frequency and the vertical axisshows intensity of a spectrum.

In this technology, a spectrum of the original signal whose frequencyband is 0≦k<FH (FIG. 1A) is expressed using a low-frequency spectrumwhose frequency band is 0≦k<FL (FIG. 1B). More specifically, thehigh-frequency spectrum (FL≦k<FH) is replaced by the low-frequencyspectrum (0≦k<FL). As a result of this processing, the spectrum as shownin FIG. 1C is obtained. Here, for simplicity of explanations, a casewith a relationship of FL=FH/2 will be explained as an example.According to information about a spectrum envelope of the originalsignal, the amplitude value of the spectrum in the high-frequency domainof the spectrum in FIG. 1C is adjusted and the spectrum as shown in FIG.1D is obtained. This is the spectrum which is the spectrum obtained byestimating the original signal.

-   Patent Document 1: National Publication of International Patent    Application No. 2001-521648 (pp. 15, FIG. 1, FIG. 2)

DISCLOSURE OF INVENTION Problems to be Solved by the Invention

Generally, spectra such as voice signal and audio signal are known tohave a harmonic structure in which a peak of spectrum appears at everyinteger multiple of a certain frequency (every predetermined pitch).This harmonic structure is important information to keep the quality ofa voice signal, audio signal or the like, and if disturbance occurs inthe harmonic structure, a listener perceives deterioration of thequality.

FIGS. 2A and 2B are diagrams illustrating problems of the conventionaltechnology.

FIG. 2A is a spectrum obtained by analyzing the spectrum of an audiosignal. As is appreciated from this figure, the original signal has aharmonic structure having an interval T on the frequency axis. On theother hand, FIG. 2B shows a spectrum obtained as a result of estimatingthe spectrum of the original signal according to the above describedtechnology. When these two spectra are compared, it is observed from thespectrum shown in FIG. 2B that the harmonic structure is maintained inlow-frequency spectrum S1 of the replacement source and high-frequencyspectrum S2 of the replacement destination, whereas the harmonicstructure is collapsed in the connection-domain (spectrum S3) betweenlow-frequency spectrum S1 and high-frequency spectrum S2.

When this estimated spectrum is converted to a time signal and listened,there is a problem that the listener perceives deterioration in qualitydue to such disturbance of the harmonic structure. This disturbance ofthe harmonic structure is caused by the fact that replacement has beenperformed with no consideration given to the shape of the harmonicstructure.

It is an object of the present invention to provide a coding apparatuscapable of coding a spectrum at a low bit rate and with high qualitywithout producing disturbance in the harmonic structure of the spectrumand a decoding apparatus capable of decoding this coded signal.

Means for Solving the Problem

The coding apparatus of the present invention adopts a configurationcomprising an acquisition section that acquires a spectrum divided intotwo bands of low-frequency band and high-frequency band, a calculationsection that calculates a parameter indicating the degree of similaritybetween the acquired spectrum of the low-frequency band and the acquiredspectrum of the high-frequency band based on the harmonic structure ofthe spectrum and a coding section that encodes the calculated parameterindicating the degree of similarity instead of the acquired spectrum ofthe high-frequency band.

The decoding apparatus of the present invention adopts a configurationcomprising a spectrum acquisition section that acquires the spectrum ofthe low-frequency band out of the spectrum divided into two bands oflow-frequency band and high-frequency band, a parameter acquisitionsection that acquires a parameter indicating the degree of similaritybetween the spectrum of the low-frequency band and the spectrum of thehigh-frequency band and a decoding section that decodes the spectra ofthe low-frequency band and high-frequency band using the acquiredspectrum of the low-frequency band and the parameter.

The coding method of the present invention comprises an acquiring stepof acquiring a spectrum divided into two bands of low-frequency band andhigh-frequency band, a calculating step of calculating a parameterindicating the degree of similarity between the acquired spectrum of thelow-frequency band and the acquired spectrum of the high-frequency bandbased on a harmonic structure of the spectrum and a coding step ofcoding the calculated parameter indicating the degree of similarityinstead of the acquired spectrum of the high-frequency band.

The decoding method of the present invention comprises a spectrumacquiring step of acquiring a spectrum of a low-frequency band out of aspectrum divided into two bands of the low-frequency band andhigh-frequency band, a parameter acquiring step of acquiring a parameterindicating the degree of similarity between the spectrum of thelow-frequency band and the spectrum of the high-frequency band and adecoding step of decoding the spectra of the low-frequency band andhigh-frequency band using the acquired spectrum of the low-frequencyband and the parameter.

Advantageous Effect of the Invention

The present invention is capable of performing coding of a spectrum at alow bit rate and with high quality without any collapse of a harmonicstructure of the spectrum. Furthermore, the present invention is alsocapable of improving sound quality when decoding this coded signal.

BRIEF DESCRIPTION OF DRAWINGS

FIG. 1 is a diagram illustrating an overview of a conventionalprocessing of replacing a spectrum of high-frequency domain by aspectrum of a low-frequency domain;

FIG. 2 is a diagram illustrating a problem of the conventionaltechnology;

FIG. 3 is a block diagram showing the principal configuration of a radiotransmission apparatus according to Embodiment 1;

FIG. 4 is a block diagram showing the internal configuration of a codingapparatus according to Embodiment 1;

FIG. 5 is a block diagram showing the internal configuration of aspectrum coding section according to Embodiment 1;

FIG. 6 is a diagram illustrating an overview of filtering processing ofa filtering section according to Embodiment 1;

FIG. 7 is a diagram illustrating how a spectrum of an estimated value ofa second spectrum changes as pitch coefficient T changes;

FIG. 8 is a diagram illustrating how a spectrum of an estimated value ofa second spectrum changes as pitch coefficient T changes;

FIG. 9 is a flow chart showing an example of a series of algorithms ofprocesses carried out by the filtering section, search section and pitchcoefficient setting section according to Embodiment 1;

FIG. 10 is a block diagram showing the principal configuration of aradio reception apparatus according to Embodiment 1;

FIG. 11 is a block diagram showing the internal configuration of adecoding apparatus according to Embodiment 1;

FIG. 12 is a block diagram showing the internal configuration of aspectrum decoding section according to Embodiment 1;

FIG. 13 is a diagram illustrating a decoded spectrum generated by afiltering section according to Embodiment 1;

FIG. 14A is a block diagram showing the principal configuration of thetransmitting side when the coding apparatus according to Embodiment 1 isapplied to a wired communication system;

FIG. 14B is a block diagram showing the principal configuration of thereceiving side when the decoding apparatus according to Embodiment 1 isapplied to a wired communication system.

FIG. 15 is a block diagram showing the principal configuration of aspectrum coding section according to Embodiment 2;

FIG. 16 is a diagram illustrating an overview of filtering using afilter according to Embodiment 2;

FIG. 17 is a block diagram showing the principal configuration of aspectrum coding section according to Embodiment 3;

FIG. 18 is a block diagram showing the principal configuration of aspectrum decoding section according to Embodiment 4; and

FIG. 19 is a block diagram showing the principal configuration of aspectrum decoding section according to Embodiment 5.

BEST MODE FOR CARRYING OUT THE INVENTION

The inventor focused attention on the characteristics such as voicesignal, audio signal or the suchlike (hereinafter, collectively referredto as “acoustic signal”), that is to say, on the fact that an acousticsignal forms a harmonic structure in the frequency axis direction,discovered the possibility of performing coding spectra of the remainingbands using spectra of some bands out of spectra of all frequency bands,and came up with the present invention.

That is, the essence of the present invention is to determine, forexample, when coding a signal spectrum divided into two frequency bandsof high-frequency domain and low-frequency domain, the degree ofsimilarity between the spectra of both the high-frequency domain andlow-frequency domain for the spectrum of the high-frequency domain andperform coding of a parameter indicating this degree of similarity.

With reference to the accompanying drawings, embodiments of the presentinvention will be explained in detail below.

Embodiment 1

FIG. 3 is a block diagram showing the principal configuration of radiotransmission apparatus 130 when a radio coding apparatus according toEmbodiment 1 of the present invention is mounted on the transmittingside of a radio communication system.

This radio transmission apparatus 130 includes coding apparatus 120,input apparatus 131, A/D conversion apparatus 132, RF modulationapparatus 133 and antenna 134.

Input apparatus 131 converts sound wave W11 audible to human ears to ananalog signal which is an electric signal and outputs the signal to A/Dconversion apparatus 132. A/D conversion apparatus 132 converts thisanalog signal to a digital signal and outputs the digital signal tocoding apparatus 120. Coding apparatus 120 encodes the input digitalsignal, generates a coded signal and outputs the coded signal to RFmodulation apparatus 133. RF modulation apparatus 133 modulates thecoded signal, generates a modulated coded signal and outputs themodulated coded signal to antenna 134. Antenna 134 transmits themodulated coded signal as radio wave W12.

FIG. 4 is a block diagram showing the internal configuration of abovedescribed coding apparatus 120. Here, a case where hierarchical coding(scalable coding) is performed will be explained as an example.

Coding apparatus 120 includes input terminal 121, downsampling section122, first layer coding section 123, first layer decoding section 124,upsampling section 125, delay section 126, spectrum coding section 100,multiplexing section 127 and output terminal 128.

A signal having an effective frequency band of 0≦k<FH is input from A/Dconversion apparatus 132 to input terminal 121. Downsampling section 122applies downsampling to the signal input via input terminal 121,generates a signal having a low sampling rate and outputs the signal.First layer coding section 123 encodes this downsampled signal, outputsthe obtained code to multiplexing section (multiplexer) 127 and alsooutputs the obtained code to first layer decoding section 124. Firstlayer decoding section 124 generates a decoded signal of a first layerbased on the code. Upsampling section 125 increases the sampling rate ofthe decoded signal of first layer coding section 123.

On the other hand, delay section 126 provides a delay of a predeterminedlength to the signal input via input terminal 121. Suppose the length ofthis delay has the same value as a time delay produced when the signalis passed through downsampling section 122, first layer coding section123, first layer decoding section 124 and upsampling section 125.Spectrum coding section 100 performs spectrum coding using the signaloutput from upsampling section 125 as a first signal and the signaloutput from delay section 126 as a second signal and outputs thegenerated code to multiplexing section 127. Multiplexing section 127multiplexes the code obtained from first layer coding section 123 withthe code obtained from spectrum coding section 100 and outputs themultiplexed parameter as an output code via output terminal 128. Thisoutput code is given to RF modulation apparatus 133.

FIG. 5 is a block diagram showing the internal configuration of abovedescribed spectrum coding section 100.

Spectrum coding section 100 includes input terminals 102, 103, frequencydomain conversion sections 104, 105, internal state setting section 106,filtering section 107, search section 108, pitch coefficient settingsection 109, filter coefficient calculation section 110 and outputterminal 111.

The first signal is input from upsampling section 125 to input terminal102. This first signal is a signal which is decoded by first layerdecoding section 124 using a coded parameter coded by first layer codingsection 123 and has an effective frequency band of 0≦k<FL. Furthermore,the second signal having an effective frequency band of 0≦k<FH (FL<FH)is input from delay section 126 to input terminal 103.

Frequency domain conversion section 104 performs frequency conversion onthe first signal input from input terminal 102 and calculates firstspectrum S1(k). On the other hand, frequency domain conversion section105 performs frequency conversion on the second signal input from inputterminal 103 and calculates second spectrum S2(k). Here, the frequencyconversion method applies a discrete Fourier transform (DFT), discretecosine transform (DCT), modified discrete cosine transform (MDCT) or thelike are used.

Internal state setting section 106 sets the internal state of a filterused in filtering section 107 using first spectrum S1(k) having aneffective frequency band of 0≦k<FL. This setting will be explained lateragain.

Pitch coefficient setting section 109 outputs pitch coefficients T tofiltering section 107 one by one while changing them little by littlewithin a predetermined search range of T_(min) to T_(max).

Filtering section 107 performs filtering of the second spectrum based onthe internal state of the filter set by internal state setting section106 and pitch coefficient T output from pitch coefficient settingsection 109 and calculates estimated value S′2(k) of the first spectrum.Details of this filtering processing which will be described later.

Search section 108 calculates a degree of similarity which is aparameter indicating similarity between second spectrum S2(k) outputfrom frequency domain conversion section 105 and estimated value S′2(k)of the second spectrum output from filtering section 107. This degree ofsimilarity will be described in detail later. Calculation processing ofthis degree of similarity is performed every time pitch coefficient T isgiven from pitch coefficient setting section 109 and pitch coefficientT′ (range of T_(min) to T_(max)) whereby the calculated degree ofsimilarity becomes a maximum is given to filter coefficient calculationsection 110.

Filter coefficient calculation section 110 calculates filter coefficientβ_(i) using pitch coefficient T′ given from search section 108 andoutputs the filter coefficient via output terminal 111. At this time,pitch coefficient T′ is also output via output terminal 111simultaneously.

Next, specific operations of the principal components of spectrum codingsection 100 will be explained in detail using mathematical expressionsbelow.

FIG. 6 illustrates an overview of filtering processing of filteringsection 107.

Here, suppose spectra of all frequency bands (0≦k<FH) are called “S(k)”for convenience and a filter function expressed by the followingequation will be used.

$\begin{matrix}{{P(z)} = \frac{1}{1 - {\sum\limits_{i = {- M}}^{M}{\beta_{i}z^{{- T} + i}}}}} & \left( {{Equation}\mspace{14mu} 1} \right)\end{matrix}$In this equation, z denotes a z conversion variable, T denotes acoefficient given from pitch coefficient setting section 109 and supposeM=1.

As shown in this figure, first spectrum S1(k) is stored in band 0≦k<FLof S(k) as the internal state of the filter. On the other hand,estimated value S′2(k) of the second spectrum obtained from thefollowing procedure is stored in band FL≦k<FH of S(k).

A spectrum expressed by the following equation (2) is substituted inS′2(k) thorough filtering processing. The substituted spectrum isobtained by adding all spectrum β_(i)·S(k−T−i), obtained by multiplyingnearby spectrums S(k−T−i) separated by i centered on the spectrum S(k−T)having a frequency lower than k by T by predetermined weighting factorβ_(i).

$\begin{matrix}{{S^{\prime}2(k)} = {\sum\limits_{i = {- 1}}^{1}{\beta_{i} \cdot {S\left( {k - T - i} \right)}}}} & \left( {{Equation}\mspace{14mu} 2} \right)\end{matrix}$At this time, suppose the input signal given to this filter is zero.That is, (Equation 2) expresses a zero input response of (Equation 1).Estimated value S′2(k) of the second spectrum in FL≦k<FH is calculatedby performing the above described calculations while changing k within arange FL≦k<FH in ascending order of frequencies (from k=FL).

The above described filtering processing is performed within rangeFL≦k<FH every time pitch coefficient T is given from pitch coefficientsetting section 109 by clearing S(k) to zero every time. That is, S(k)is calculated every time pitch coefficient T changes and output tosearch section 108.

Next, calculation processing of the degree of similarity performed bysearch section 108 and derivation processing of optimum pitchcoefficient T will be explained.

First, there are various definitions of the degree of similarity. Here,a case where the degree of similarity defined by the following equationbased on a least square error method is used assuming that filtercoefficients β⁻¹ and β₁ are 0 will be explained as an example.

$\begin{matrix}{E = {\sum\limits_{k = {FL}}^{{FH} - 1}{S\; 2(k)^{2}\frac{\left( {\sum\limits_{k = {FL}}^{{FH} - 1}{S\; 2{(k) \cdot S^{\prime}}2(k)}} \right)^{2}}{\sum\limits_{k = {FL}}^{{FH} - 1}{S^{\prime}2(k)^{2}}}}}} & \left( {{Equation}\mspace{14mu} 3} \right)\end{matrix}$In the case where this degree of similarity is used, filter coefficientβ_(i) is determined after optimum pitch coefficient T is calculated.Here, E denotes a square error between S2(k) and S′2(k). In thisequation, the first term of the right side becomes a fixed value whichis irrelevant to pitch coefficient T, and therefore pitch coefficient Tfor generating S′2(k) which makes a maximum of the second term of theright side is searched. The second term of the right side of thisequation will be called a “degree of similarity.”

FIG. 7A to FIG. 7E are diagrams illustrating how the spectrum ofestimated value S′2(k) of the second spectrum changes as pitchcoefficient T changes.

FIG. 7A is a diagram illustrating the first spectrum having a harmonicstructure stored as an internal state. Furthermore, FIG. 7B to FIG. 7Dare diagrams illustrating spectra of estimated values S′2(k) of thesecond spectrum calculated by performing filtering using three types ofpitch coefficients T₀, T₁, T₂. FIG. 7E shows second spectrum S2(k) to becompared with the spectrum of estimated value S′2(k).

In the example shown in this figure, the spectrum shown in FIG. 7C issimilar to the spectrum shown in FIG. 7E, and therefore it is realizedthat the degree of similarity calculated using T₁ shows the highestvalue. That is, T₁ is an optimum value as pitch coefficient T wherebythe harmonic structure can be maintained.

FIG. 8A to FIG. 8E domain also figures similar to FIG. 7A to FIG. 7E,but here the phase of the first spectrum stored as the internal state isdifferent from that of FIG. 7A to FIG. 7E. However, in the example shownin this figure, pitch coefficient T whereby the harmonic structure ismaintained is also T₁.

Thus, changing pitch coefficient T and finding T of a maximum degree ofsimilarity is equivalent to finding out a pitch (or an integer multiplethereof) of the harmonic structure of the spectrum on a try-and-errorbasis. The coding apparatus of this embodiment calculates estimatedvalue S′2(k) of the second spectrum based on the pitch of this harmonicstructure, and therefore the harmonic structure does not collapse in theconnection area between the first spectrum and estimated spectrum. Thisis easily understandable considering that estimated value S′2(k) of theconnection section when k=FL is calculated based on the first spectrumseparated by pitch (or an integer multiple thereof) T of the harmonicstructure.

Furthermore, pitch coefficient T expresses an integer multiple (integervalue) of the frequency interval of the spectrum data. However, thepitch of the actual harmonic structure is often a non-integer value.Therefore, by selecting appropriate weighting factor β_(i) and applyinga weighted addition to M neighboring data centered on T, it is possibleto express a pitch of the harmonic structure of a non-integer valuewithin a range from T−M to T+M.

FIG. 9 is a flow chart showing an example of a series of algorithms ofprocesses performed by filtering section 107, search section 108 andpitch coefficient setting section 109. An overview of these processeshas already been explained, and therefore detailed explanations of theflow will be omitted.

Next, the calculation processing of a filter coefficient by filtercoefficient calculation section 110 will be explained.

Filter coefficient calculation section 110 determines filter coefficientβ_(i) that minimizes square distortion E in the following equation usingpitch coefficient T′ given from search section 108.

$\begin{matrix}{E = {\sum\limits_{k = {FL}}^{{FH} - 1}\left( {{S\; 2(k)} - {\sum\limits_{i = {- 1}}^{1}{\beta_{i}{S\left( {k - T^{\prime} - i} \right)}}}} \right)^{2}}} & \left( {{Equation}\mspace{14mu} 4} \right)\end{matrix}$Filter coefficient calculation section 110 holds a combination of aplurality of βi (i=−1,0,1) as a data table beforehand, determines acombination of βi (i=−1,0,1) that minimizes square distortion E of abovedescribed (Equation 4) and outputs an index thereof.

Thus, for the spectrum of an input signal divided into two parts of alow-frequency domain (0≦k<FL) and high-frequency domain (FL≦k<FH), thecoding apparatus of this embodiment estimates the shape of thehigh-frequency spectrum using filtering section 107 that includes thelow-frequency spectrum as the internal state, encodes and outputs aparameter indicating the filter characteristic of filtering section 107instead of the high-frequency spectrum, and therefore, it is possible toperform coding of the spectrum at a low bit rate and with high quality.

Furthermore, in the above described configuration, when filteringsection 107 estimates the shape of the high-frequency spectrum using thelow-frequency spectrum, pitch coefficient setting section 109 changesthe frequency difference between the low-frequency spectrum which servesas a reference for estimation and the high-frequency spectrum, that is,pitch coefficient T, in various ways and outputs the frequencydifference, and search section 108 detects T corresponding to a maximumdegree of similarity between the low-frequency spectrum andhigh-frequency spectrum. Therefore, it is possible to estimate the shapeof the high-frequency spectrum based on the pitch of the harmonicstructure of the overall spectrum and perform coding while maintainingthe harmonic structure of the overall spectrum.

Furthermore, there is no need for setting the bandwidth of thelow-frequency spectrum based on the pitch of the harmonic structure.That is, it is not necessary to match the bandwidth of the low-frequencyspectrum to the pitch of the harmonic structure (or an integer multiplethereof), and it is possible to set a bandwidth arbitrarily. This isbecause the above described configuration allows spectra to be connectedsmoothly in the connection section between the low-frequency spectrumand high-frequency spectrum without matching the bandwidth of thelow-frequency spectrum to the pitch of the harmonic structure.

This embodiment has explained the case where M=1 in (Equation 1) as anexample, but M is not limited to this and an integer (natural number) of0 or greater can also be used.

Furthermore, this embodiment has explained the coding apparatus thatperforms hierarchical coding (scalable coding) as an example, but abovedescribed spectrum coding section 100 can also be mounted on a codingapparatus that performs coding based on other schemes.

Furthermore, this embodiment has explained the case where spectrumcoding section 100 includes frequency domain conversion sections 104,105. These are components necessary when a time domain signal is used asan input signal, but the frequency domain conversion section is notnecessary in a structure in which the spectrum is directly input tospectrum coding section 100.

Furthermore, this embodiment has explained the case where thehigh-frequency spectrum is coded using the low-frequency spectrum, thatis, using the low-frequency spectrum as a reference for coding, but themethod of setting the spectrum which serves as a reference is notlimited to this, and it is also possible to perform coding of thelow-frequency spectrum using the high-frequency spectrum or performcoding of the spectra of other regions using the spectrum of anintermediate frequency band as a reference for coding though these arenot desirable from the standpoint of effectively using energy.

FIG. 10 is a block diagram showing the principal configuration of radioreception apparatus 180 that receives a signal transmitted from radiotransmission apparatus 130.

This radio reception apparatus 180 includes antenna 181, RF demodulationapparatus 182, decoding apparatus 170, D/A conversion apparatus 183 andoutput apparatus 184.

Antenna 181 receives a digital coded acoustic signal as radio wave W12,generates a digital received coded acoustic signal which is an electricsignal and provides it to RF demodulation apparatus 182. RF demodulationapparatus 182 demodulates the received coded acoustic signal fromantenna 181, generates the demodulated coded acoustic signal andprovides it to decoding apparatus 170.

Decoding apparatus 170 receives the digital demodulated coded acousticsignal from RF demodulation apparatus 182, performs decoding processing,generates a digital decoded acoustic signal and provides it to D/Aconversion apparatus 183. D/A conversion apparatus 183 converts thedigital decoded voice signal from decoding apparatus 170, generates ananalog decoded voice signal and provides it to output apparatus 184.Output apparatus 184 converts the analog decoded voice signal which isan electric signal to air vibration and outputs it as sound wave W13 soas to be audible to human ears.

FIG. 11 is a block diagram showing the internal configuration of abovedescribed decoding apparatus 170. Here, a case where a signal subjectedto hierarchical coding is decoded will be explained as an example.

This decoding apparatus 170 includes input terminal 171, separationsection 172, first layer decoding section 173, upsampling section 174,spectrum decoding section 150 and output terminals 176, 177.

RF demodulation apparatus 182 inputs digital demodulated coded acousticsignal to input terminal 171. Separation section 172 separates thedemodulated coded acoustic signal input via input terminal 171 andgenerates a code for first layer decoding section 173 and a code forspectrum decoding section 150. First layer decoding section 173 decodesthe decoded signal having signal band 0≦k<FL using the code obtainedfrom separation section 172 and provides this decoded signal toupsampling section 174. Furthermore, the other output is connected tooutput terminal 176. This allows, when the first layer decoded signalgenerated by first layer decoding section 173 needs to be output, thefirst layer decoded signal can be output via this output terminal 176.

Upsampling section 174 increases the sampling frequency of the firstlayer decoded signal provided from first layer decoding section 173.Spectrum decoding section 150 is given the code separated by separationsection 172 and the upsampled first layer decoded signal generated byupsampling section 174. Spectrum decoding section 150 performs spectrumdecoding which will be described later, generates a decoded signalhaving signal band 0≦k<FH and outputs the decoded signal via outputterminal 177. Spectrum decoding section 150 regards the upsampled firstlayer decoded signal provided from upsampling section 174 as the firstsignal and performs processing.

According to this configuration, when the first layer decoded signalgenerated by first layer decoding section 173 needs to be output, thefirst layer decoded signal can be output from output terminal 176.Furthermore, when an output signal of higher quality of spectrumdecoding section 150 needs to be output, the output signal can be outputfrom output terminal 177. Decoding apparatus 170 outputs either one ofsignals output from terminal 176 or output terminal 177 and provides thesignal to D/A conversion apparatus 183. Which signal is to be outputdepends on the setting of the application or judgment of the user.

FIG. 12 is a block diagram showing the internal configuration of abovedescribed spectrum decoding section 150.

This spectrum decoding section 150 includes input terminals 152, 153,frequency domain conversion section 154, internal state setting section155, filtering section 156, time domain conversion section 158 andoutput terminal 159.

A filter coefficient indicating a code obtained by spectrum codingsection 100 is input to input terminal 152 via separation section 172.Furthermore, a first signal having an effective frequency band of 0≦k<FLis input to input terminal 153. This first signal is the first layerdecoded signal decoded by first layer decoding section 173 and upsampledby upsampling section 174.

Frequency domain conversion section 154 converts the frequency of thetime domain signal input from input terminal 153 and calculates firstspectrum S1(k). As the frequency conversion method, a discrete Fouriertransform (DFT), discrete cosine transform (DCT), modified discretecosine transform (MDCT) or the like is used.

Internal state setting section 155 sets the internal state of a filterused in filtering section 156 using first spectrum S1(k).

Filtering section 156 performs filtering of the first spectrum based onthe internal state of the filter set by internal state setting section155 and pitch coefficient T′ and filter coefficient β provided frominput terminal 152 and calculates estimated value S′2(k) of the secondspectrum. In this case, filtering section 156 uses the filter functiondescribed in (Equation 1).

Time domain conversion section 158 converts decoded spectrum S′(k)obtained from filtering section 156 to a time domain signal and outputsthe decoded spectrum via output terminal 159. Here, processing such asappropriate windowing and overlapped addition is performed as requiredto avoid discontinuation that may occur between frames.

FIG. 13 shows decoded spectrum S′(k) generated by filtering section 156.

As shown in this figure, decoded spectrum S′(k) having frequency band0≦k<FL consists of first spectrum S1(k) and decoded spectrum S′(k)having frequency band FL≦k<FH consists of estimated value S′2(k) of thesecond spectrum.

Thus, the decoding apparatus of this embodiment has the configurationcorresponding to the coding method according to this embodiment, andtherefore, it is possible to decode a coded acoustic signal efficientlywith fewer bits and output an acoustic signal of high quality.

Here, the case where the coding apparatus or decoding apparatusaccording to this embodiment is applied to a radio communication systemhas been explained as an example, but the coding apparatus or decodingapparatus according to this embodiment is also applicable to a wiredcommunication system as shown below.

FIG. 14A is a block diagram showing the principal configuration of thetransmitting side when the coding apparatus according to this embodimentis applied to a wired communication system. The same components as thoseshown in FIG. 3 are assigned the same reference numerals andexplanations thereof will be omitted.

Wired transmission apparatus 140 includes coding apparatus 120, inputapparatus 131 and A/D conversion apparatus 132 and an output thereof isconnected to network N1.

The input terminal of A/D conversion apparatus 132 is connected to theoutput terminal of input apparatus 131. The input terminal of codingapparatus 120 is connected to the output terminal of A/D conversionapparatus 132. The output terminal of coding apparatus 120 is connectedto network N1.

Input apparatus 131 converts sound wave W11 audible to human ears to ananalog signal which is an electric signal and provides it to A/Dconversion apparatus 132. A/D conversion apparatus 132 converts theanalog signal to a digital signal and provides the digital signal tocoding apparatus 120. Coding apparatus 120 encodes the input digitalsignal, generates a code and outputs the code to network N1.

FIG. 14B is a block diagram showing the principal configuration of thereceiving side when the decoding apparatus according to this embodimentis applied to a wired communication system. The same components as thoseshown in FIG. 10 are assigned the same reference numerals andexplanations thereof will be omitted.

Wired reception apparatus 190 includes reception apparatus 191 connectedto network N1, decoding apparatus 170, D/A conversion apparatus 183 andoutput apparatus 184.

The input terminal of reception apparatus 191 is connected to networkN1. The input terminal of decoding apparatus 170 is connected to theoutput terminal of reception apparatus 191. The input terminal of D/Aconversion apparatus 183 is connected to the output terminal of decodingapparatus 170. The input terminal of output apparatus 184 is connectedto the output terminal of D/A conversion apparatus 183.

Reception apparatus 191 receives a digital coded acoustic signal fromnetwork N1, generates a digital received acoustic signal and providesthe signal to decoding apparatus 170. Decoding apparatus 170 receivesthe received acoustic signal from reception apparatus 191, performsdecoding processing on this received acoustic signal, generates adigital decoded acoustic signal and provides it to D/A conversionapparatus 183. D/A conversion apparatus 183 converts the digital decodedvoice signal from decoding apparatus 170, generates an analog decodedvoice signal and provides it to output apparatus 184. Output apparatus184 converts the analog decoded acoustic signal which is an electricsignal to air vibration and outputs it as sound wave W13 audible tohuman ears.

Thus, according to the above described configuration, it is possible toprovide a wired transmission/reception apparatus having operations andeffects similar to those of the above described radiotransmission/reception apparatus.

Embodiment 2

FIG. 15 is a block diagram showing the principal configuration ofspectrum coding section 200 in a coding apparatus according toEmbodiment 2 of the present invention. This spectrum coding section 200has a basic configuration similar to that of spectrum coding section 100shown in FIG. 5 and the same components are assigned the same referencenumerals and explanations thereof will be omitted.

A feature of this embodiment is to make a filter function used in thefiltering section simpler than that in Embodiment 1.

For the filter function used in filtering section 201, a simplified oneas shown in the following equation is used.

$\begin{matrix}{{P(z)} = \frac{1}{1 - z^{- T}}} & \left( {{Equation}\mspace{14mu} 5} \right)\end{matrix}$This equation corresponds to a filter function assuming M=0, β₀=1 in(Equation 1).

FIG. 16 illustrates an overview of filtering using the above describedfilter.

Estimated value S′2(k) of a second spectrum is obtained by sequentiallycopying low-frequency spectra separated by T. Furthermore, searchsection 108 determines optimum pitch coefficient T′ by searching forpitch coefficient T which minimizes E of (Equation 3) as in the case ofEmbodiment 1. Pitch coefficient T′ obtained in this way is output viaoutput terminal 111. In this configuration, the characteristic of thefilter is determined only by pitch coefficient T.

Note that the filter of this embodiment is characterized in that itoperates in a way similar to an adaptive codebook, one of components ofa CELP (Code-Excited Linear Prediction) scheme which is a representativetechnology of low-rate voice coding.

Next, the spectrum decoding section that decodes a signal coded by abovedescribed spectrum coding section 200 will be explained (not shown).

This spectrum decoding section has a configuration similar to that ofspectrum decoding section 150 shown in FIG. 12, and therefore detailedexplanations thereof will be omitted, and it has the following features.That is, when filtering section 156 calculates estimated value S′2(k) ofthe second spectrum, it uses the filter function described in (Equation5) instead of the filter function described in (Equation 1). It is onlypitch coefficient T′ that is provided from input terminal 152. That is,which of the filter function described in (Equation 1) or (Equation 5)should be used is determined depending on the type of the filterfunction used on the coding side and the same filter function used onthe coding side is used.

Thus, according to this embodiment, the filter function used in thefiltering section is made simpler, which result in eliminating thenecessity for installing a filter coefficient calculation section.Therefore, it is possible to estimate the second spectrum(high-frequency spectrum) with a smaller amount of calculation and alsoreduce the circuit scale.

Embodiment 3

FIG. 17 is a block diagram showing the principal configuration ofspectrum coding section 300 in a coding apparatus according toEmbodiment 3 of the present invention. This spectrum coding section 300has a basic configuration similar to that of spectrum coding section 100shown in FIG. 5 and the same components are assigned the same referencenumerals and explanations thereof will be omitted.

A feature of this embodiment is to further comprise outline calculationsection 301 and multiplexing section 302 and perform coding of envelopeinformation about a second spectrum after estimating the secondspectrum.

Search section 108 outputs optimum pitch coefficient T′ to multiplexingsection 302 and outputs estimated value S′2(k) of the second spectrumgenerated using this pitch coefficient T′ to outline calculation section301. Outline calculation section 301 calculates envelope informationabout second spectrum S2(k) based on second spectrum S2(k) provided fromfrequency domain conversion section 105. Here, a case where thisenvelope information is expressed by spectrum power for each subband andfrequency band FL≦k<FH is divided into J subbands will be explained asan example. At this time, the spectrum power of the jth subband isexpressed by the following equation.

$\begin{matrix}{{B(j)} = {\sum\limits_{k = {{BL}{(j)}}}^{{BH}{(j)}}{S\; 2(k)^{2}}}} & \left( {{Equation}\mspace{14mu} 6} \right)\end{matrix}$In this equation, BL(j) denotes a minimum frequency of the j^(th)subband, BH(j) denotes a maximum frequency of the j^(th) subband. Thesubband information of the second spectrum obtained in this way isregarded as the spectrum envelope information about the second spectrum.

In a similar fashion, subband information B′(j) of estimated valueS′2(k) on the second spectrum is calculated according to the followingequation,

$\begin{matrix}{{B^{\prime}(j)} = {\sum\limits_{k = {{BL}{(j)}}}^{{BH}{(j)}}{S^{\prime}2(k)^{2}}}} & \left( {{Equation}\mspace{14mu} 7} \right)\end{matrix}$and amount of variation V(j) for each subband is calculated according tothe following equation.

$\begin{matrix}{{V(j)} = \sqrt{\frac{B(j)}{B^{\prime}(j)}}} & \left( {{Equation}\mspace{14mu} 8} \right)\end{matrix}$

Next, outline calculation section 301 encodes amount of variation V(j),obtains the coded amount of variation V(j) and outputs the index thereofto multiplexing section 302. Multiplexing section 302 multiplexesoptimum pitch coefficient T′ obtained from search section 108 and anindex of amount of variation V(j) output from outline calculationsection 301 and outputs the multiplexing result via output terminal 111.

Thus, this embodiment makes it possible to improve an accuracy of theestimated value of the high-frequency spectrum since the envelopeinformation about the high-frequency spectrum is further coded after ahigh-frequency spectrum is estimated.

Embodiment 4

FIG. 18 is a block diagram showing the principal configuration ofspectrum decoding section 550 according to Embodiment 4 of the presentinvention. This spectrum decoding section 550 has a basic configurationsimilar to that of spectrum decoding section 150 shown in FIG. 12, andtherefore the same components are assigned the same reference numeralsand explanations thereof will be omitted.

A feature of this embodiment is to further comprise separation section551, spectrum envelope decoding section 552 and spectrum adjustingsection 553. This allows spectrum coding section 300 or the like shownin Embodiment 3 to perform decoding of a code resulting from coding ofenvelope information as well as coding of an estimated spectrum of ahigh-frequency spectrum.

Separation section 551 separates a code input via input terminal 152,provides information about a filtering coefficient to filtering section156 and provides information about a spectrum envelope to spectrumenvelope decoding section 552.

Spectrum envelope decoding section 552 decodes amount of variationV_(q)(j) obtained by coding amount of variation V(j) from the spectrumenvelope information given from separation section 551.

Spectrum adjusting section 553 multiplies decoded spectrum S′(k)obtained from filtering section 156 by decoded amount of variationV_(q)(j) for each subband obtained from spectrum envelop decodingsection 552 according to the following equation,S3(k)=S′(k)·V _(q)(j)(BL(j)≦k≦BH(j), for all j)  (Equation 9)adjusts a spectral shape in frequency band FL≦k<FH of decoded spectrumS′(k) and generates adjusted decoded spectrum S3(k). This adjusteddecoded spectrum S3(k) is output to time domain conversion section 158and converted to a time domain signal.

Thus, according to this embodiment, it is possible to decode a codeincluding envelope information.

This embodiment has explained the case where the spectrum envelopeinformation provided from separation section 551 is value V_(q)(j)obtained by coding amount of variation V(j) for each subband shown in(Equation 8) as an example, but the spectrum envelope information is notlimited to this.

Embodiment 5

FIG. 19 is a block diagram showing the principal configuration of aspectrum decoding section 650 in a decoding apparatus according toEmbodiment 5 of the present invention. This spectrum decoding section650 has a basic configuration similar to that of spectrum decodingsection 550 shown in FIG. 18, and therefore the same components areassigned the same reference numerals and explanations thereof will beomitted.

A feature of this embodiment is to further comprise LPC spectrumcalculation section 652, use an LPC spectrum calculated with an LPCcoefficient as spectrum envelope information, estimate a secondspectrum, and then multiply the second spectrum by the LPC spectrum toobtain a more accurate estimated value of the second spectrum.

LPC spectrum calculation section 652 calculates LPC spectrum env(k) fromLPC coefficient α(j) input via input terminal 651 according to thefollowing equation.

$\begin{matrix}{{{env}(k)} = {\frac{1}{1 - {\sum\limits_{j = 1}^{NP}{{\alpha(j)}{\mathbb{e}}^{{- j}\frac{2\pi\; j\; k}{FH}}}}}}} & \left( {{Equation}\mspace{14mu} 10} \right)\end{matrix}$Here, NP denotes the order of the LPC coefficient. Furthermore, it isalso possible to calculate LPC spectrum env(k) using variable γ(0<γ<1)and changing the characteristic of the LPC spectrum.

In this case, LPC spectrum env(k) is expressed by the followingequation.

$\begin{matrix}{{{env}(k)} = {\frac{1}{1 - {\sum\limits_{j = 1}^{NP}{{\alpha(j)} \cdot \gamma^{j} \cdot {\mathbb{e}}^{{- j}\frac{2\pi\; j\; k}{FH}}}}}}} & \left( {{Equation}\mspace{14mu} 11} \right)\end{matrix}$Here, γ may be defined as a fixed value or may also take a value whichis variable from one frame to another. LPC spectrum env(k) calculated inthis way is output to spectrum adjusting section 553.

Spectrum adjusting section 553 multiplies decoded spectrum S′(k)obtained from filtering section 156 by LPC spectrum env(k) obtained fromLPC spectrum calculation section 652 according to the followingequation,S3(k)=S′(k)·env(k)(FL≦k<FH)  (Equation 12)adjusts the spectrum in frequency band FL≦k<FH of decoded spectrum S′(k)and generates adjusted decoded spectrum S3(k). This adjusted decodedspectrum S3(k) is provided to time domain conversion section 158 andconverted to a time domain signal.

Thus, according to this embodiment, using an LPC spectrum as spectrumenvelope information makes it possible to obtain a more accurateestimated value of the second spectrum.

The coding apparatus or decoding apparatus according to the presentinvention can be mounted on a communication terminal apparatus and basestation apparatus in a mobile communication system, and therefore, it ispossible to provide a communication terminal apparatus and base stationapparatus having operations and effects similar to those describedabove.

The case where the present invention is constructed by hardware has beenexplained as an example so far, but the present invention can also beimplemented by software.

The present application is based on Japanese Patent Application No.2003-323658 filed on Sep. 16, 2003, entire content of which is expresslyincorporated by reference herein.

INDUSTRIAL APPLICABILITY

The coding apparatus and decoding apparatus according to the presentinvention have the effect of performing coding at a low bit rate and isalso applicable to a radio communication system or the like.

1. A spectrum coding apparatus that encodes a spectrum including a firstband and a second band, the spectrum coding apparatus comprising: ageneration section that calculates an estimated spectrum of the secondband using pitch information on a harmonic structure of the first bandspectrum having at least a bandwidth of the second band; an acquisitionsection that determines the pitch information on the harmonic structureof the first band spectrum that minimizes a distortion measure betweenthe estimated spectrum of the second band and the spectrum of the secondband; and a coding section that encodes the determined pitch informationon the harmonic structure of the first band spectrum that minimizes thedistortion measure, instead of the spectrum of said second band.
 2. Thespectrum coding apparatus according to claim 1, wherein the acquisitionsection acquires information about the spectrum of the first band beingshape information about a spectrum similar to the spectrum of the secondband.
 3. The spectrum coding apparatus according to claim 1, wherein theacquisition section acquires a filter parameter related to the estimatedspectrum of the second band as information about the spectrum of thefirst band.
 4. The coding apparatus according to claim 1, wherein: thefirst band is a low frequency band lower than a predetermined threshold;the second band is a high frequency band equal to or higher than thepredetermined threshold; the generation section generates an estimatedspectrum of the high frequency band using a filter comprising a spectrumof the low frequency band as an internal state; the acquisition sectionacquires a filter parameter that minimizes the distortion measurebetween an estimated spectrum of the high frequency band and a spectrumof the high frequency band; and the coding section encodes the filterparameter that minimizes the distortion measure, instead of the spectrumof the high frequency band.
 5. The coding apparatus according to claim4, wherein the generation section generates the estimated spectrum ofthe high frequency band using the filter comprising the spectrum of thelow frequency band separated by an integer multiple of a pitch of theharmonic structure from the spectrum of the high frequency band as theinternal state.
 6. The coding apparatus according to claim 5, whereinthe filter parameter is a filter coefficient.
 7. The coding apparatusaccording to claim 4, wherein the coding section also encodes envelopeinformation about the spectrum of the high frequency band.
 8. The codingapparatus according to claim 4, wherein the coding section also encodesinformation about a power ratio between the spectrum of the lowfrequency band and the spectrum of the high frequency band.
 9. Thecoding apparatus according to claim 4, wherein a filter fimetion of thefilter is expressed by an equation as follows and the generation sectiongenerates the estimated spectrum using a zero input response of thefilter:${P(z)} = \frac{1}{1 - {\sum\limits_{i = {- M}}^{M}\;{\beta_{i}z^{{- T} + i}}}}$where P(z): Filter function z: z conversion variable 2M+1: Order offilter β: Weighting factor T: Pitch coefficient.
 10. The codingapparatus according to claim 9, wherein M=0 and β0=1 in the filterfunction.
 11. The coding apparatus according to claim 4, wherein thespectrum of the low frequency band is obtained from a signal decodedafter being coded in a lower layer in hierarchical coding.
 12. Thecoding apparatus according to claim 4, wherein: the generation sectioncalculates an estimated value of the high frequency band using one of aplurality of parameter candidates prepared in advance, the acquisitionsection calculates a distortion measure between the spectrum of the highfrequency band and the estimated value, selects a parameter candidatethat minimizes the distortion measure, and acquires information thatspecifies the selected parameter candidate as the filter parameter. 13.The coding apparatus according to claim 4, wherein: the generationsection performs filtering using one of a plurality of pitch coefficientcandidates prepared in advance, calculates an estimated value of thespectrum of the high frequency band, the acquisition section calculatesa distortion measure between the spectrum of the high frequency band andthe estimated value, selects a pitch coefficient candidate thatminimizes the distortion measure, and acquires information thatspecifies the selected pitch coefficient candidate as the filterparameter.
 14. A communication terminal apparatus comprising the codingapparatus according to claim
 4. 15. A scalable coding apparatus thatencodes a voice signal or audio signal separated into a low frequencyband and high frequency band, the scalable coding apparatus comprising:a first coding section that encodes a low frequency band signal of thevoice signal or the audio signal; and a second coding section thatencodes a high frequency band signal of the voice signal or the audiosignal using the low frequency band signal, wherein: the second codingsection comprises: a first spectrum generation section that performsfrequency domain conversion on the low-frequency band signal andgenerates a spectrum of the low frequency band; a second spectrumgeneration section that performs frequency domain conversion on thevoice signal or the audio signal and generates a spectrum having the lowfrequency band and high frequency band; the coding apparatus accordingto claim 4, and the acquisition section in the coding apparatus of claim4 acquires the spectra generated by the first and second spectrumgeneration sections.
 16. A communication terminal apparatus comprisingthe scalable coding apparatus according to claim
 15. 17. A decodingapparatus comprising: a spectrum acquisition section that acquires aspectrum of a low frequency band out of a spectrum having a lowfrequency band and a high frequency band; a parameter acquisitionsection that acquires pitch information on a harmonic structure of thelow frequency band spectrum that minimizes a distortion measure betweenan estimated spectrum of the high frequency band and the spectrum of thehigh frequency band, the estimated spectrum of the high frequency bandbeing calculated at a corresponding encoding apparatus; and a decodingsection that decodes the spectrum of the low frequency band and thespectrum of the high frequency band using the spectrum of the lowfrequency band and the pitch information on the harmonic structure ofthe low frequency band spectrum.
 18. The decoding apparatus according toclaim 17, further comprising: an envelope information acquisitionsection that acquires envelope information about the spectrum of thehigh frequency band, wherein the decoding section performs the decodingusing also the envelope information.
 19. The decoding apparatusaccording to claim 17, wherein: the decoding section comprises anestimation section that estimates the spectrum of the high frequencyband using a filter comprising the spectrum of the low frequency band asan internal state and the parameter as a filter characteristic, a filterfunction of the filter is expressed by an equation as follows; and theestimation section performs the estimation using a zero input responseof the filter:${P(z)} = \frac{1}{1 - {\sum\limits_{i = {- M}}^{M}\;{\beta_{i}z^{{- T} + i}}}}$where P(z): Filler function z: z conversion variable 2M+1: Order offilter β: Weighting factor T: Pitch coefficient.
 20. The decodingapparatus according to claim 19, wherein M=0 and β0=1 in the filterfunction.
 21. The decoding apparatus according to claim 17, wherein thespectrum of the low frequency band is generated from a signal decoded ina lower layer in hierarchical coding.
 22. The decoding apparatusaccording to claim 17, wherein: the decoding section comprises anestimation section that estimates the spectrum of the high frequencyband using a filter comprising the spectrum of the low frequency band asan internal state and the parameter as a filter characteristic; theparameter comprises a pitch coefficient; and the estimation sectionperforms filtering using the pitch coefficient and calculates anestimated value of the spectrum of the high frequency band.
 23. Acommunication terminal apparatus comprising the decoding apparatusaccording to claim
 17. 24. A coding method for encoding a spectrumincluding a first band and a second band, the coding method comprisingthe steps of: calculating, by a calculation device, an estimatedspectrum of the second band using pitch information on a harmonicstructure of the first band spectrum having at least a bandwidth of thesecond band; determining the pitch information on the harmonic structureof the first band spectrum that minimizes a distortion measure betweenthe estimated spectrum of the second band and the spectrum of the secondband; and encoding the determined pitch information on the harmonicstructure of the first band spectrum that minimizes the distortionmeasure, instead of the spectrum of the second band.
 25. A decodingmethod comprising the steps of: acquiring a spectrum of a low frequencyband out of a spectrum having a low frequency band and a high frequencyband; acquiring pitch information on a harmonic structure of the lowfrequency band spectrum that minimizes a distortion measure between anestimated spectrum of the high frequency band and the spectrum of thehigh frequency band, the estimated spectrum of the high frequency bandbeing calculated at a corresponding encoding apparatus; and decoding, bya decoder, the spectrum of the low frequency band and the spectrum ofthe high frequency band using the spectrum of the low frequency band andthe pitch information on the harmonic structure of the low frequencyband-spectrum.